From ee6f91968c6bae25daed14a3f16e06e478c9d5dd Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Wed, 29 Aug 2007 14:16:07 +0000 Subject: [PATCH] Merged revisions 81331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4 lines (closes issue #9690) Reported by: mattv Make rtp timeouts work even if two RTP streams are directly bridged in the RTP stack. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 76f734c8ed..b28cb0ba7c 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -16427,13 +16427,10 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t) usleep(1); sip_pvt_lock(dialog); } - if (!(ast_rtp_get_bridged(dialog->rtp))) { - ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n", - dialog->owner->name, (long) (t - dialog->lastrtprx)); - /* Issue a softhangup */ - ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV); - } else - ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", dialog->owner->name, (long) (t - dialog->lastrtprx)); + ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n", + dialog->owner->name, (long) (t - dialog->lastrtprx)); + /* Issue a softhangup */ + ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV); ast_channel_unlock(dialog->owner); /* forget the timeouts for this call, since a hangup has already been requested and we don't want to